86 results about "Latency (audio)" patented technology

An electronic device can implement a zero-latency digital assistant by capturing audio input from a microphone and using a first processor to write audio data representing the captured audio input to a memory buffer. In response to detecting a user input while capturing the audio input, the device can determine whether the user input meets a predetermined criteria. If the user input meets the criteria, the device can use a second processor to identify and execute a task based on at least a portion of the contents of the memory buffer.

The present disclosure is related to transmitting and receiving media channels, such as audio and video channels. These channels may be transmitted as packets from one or more transmitting devices to one or more receiving devices for playout. Certain embodiments of the present disclosure include systems, methods, and computer- readable media for determining latency of a data network for synchronised playout of received signals. Additionally, certain embodiments of the present disclosure include systems, methods, and computer-readable media for synchronising playout among devices connected to a data network.

The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.

The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.

The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.

This invention relates generally to a packet recovery algorithm for real-time (live) multi-media communication over packet-switched networks, such as the Internet. Such multi-media communication includes video, audio, data or any combination thereof. More specifically, the invention comprises a forward error correction (FEC) algorithm that addresses both random and burst packet loss and errors, and that can be adjusted to tradeoff the recoverability of missing packets and the latency incurred. The transmitter calculates parity packets for the rows, columns and diagonals of a block of multi-media data packets using the exclusive or (XOR) operation and communicates the parity packets along with the multi-media data packets to the receiver. The receiver uses the parity packets to recover missing multi-media data packets in the block. The FEC algorithm is designed to be able to recover long bursts of consecutive missing data packets. If some parity packets are missing, they too can be recovered using an extra single parity packet, so that they can be used to recover other missing data packets. The invention applies to both one-way real-time streaming applications and two-way real-time interactive applications, and to both wired and wireless networks. The invention retains backwards compatibility with existing standards governing FEC for professional video over IP networks.

The present invention is a method and a computer program for allowing individual artists to produce a latency corrected collaborative audio work by providing a first artist the opportunity to create a base track which may then be accessed by additional artists who can add audio tracks which harmonize with the base track. Artists may add synchronized tracks to the base track at any time from a remote location thereby allowing artists in different geographical locations to produce a collaborative work of art simultaneously or over an extended time period. The present invention includes a novel method for latency correction, which corrects for delays caused by various types of user equipment, and ensures each track is properly synchronized with the others. The present invention also includes a novel interface where a slot machine or slider user interface allows for ease of use from a browser.

A method of delivering audio to one or more client computing devices includes receiving, by an audio server computing device, a live audio signal starting at a first time; processing, by the audio server computing device, the live audio signal, thereby creating a data representation of the live audio signal; transmitting, by the audio server computing device, via a wireless network in electronic communication with the audio server computing device, the data representation of the live audio signal to the one or more client computing devices; interpreting, by the one or more client computing devices, the data representation of the live audio signal, thereby producing an interpreted audio signal; and providing, by the one or more client computing devices, the interpreted audio signal to a user listening device starting at a second time. A latency between the first time and the second time is less than 100 milliseconds.

A method for transmission and low-latency real-time output and / or processing of an audio data stream that is transmitted from at least one transmitter to at least one receiver over a jittering transmission path. The method includes a calibration for determining a distribution of latencies in transmission of packets of the audio data stream, whereby a group of packets of the audio data stream is used as calibration packets and wherein a reference time grid and an offset of a fastest calibration packet are determined. Then, a shift of an output time grid for audio output and / or processing, based on the reference time grid and the determined offset of the fastest calibration packet, and the audio packets of the audio data stream are provided according to the output time grid for audio output and / or processing.

An electronic device can implement a zero-latency digital assistant by capturing audio input from a microphone and using a first processor to write audio data representing the captured audio input to a memory buffer. In response to detecting a user input while capturing the audio input, the device can determine whether the user input meets a predetermined criteria. If the user input meets the criteria, the device can use a second processor to identify and execute a task based on at least a portion of the contents of the memory buffer.